1. Field of the Invention
The present invention relates generally to a method and apparatus for carrying real time services, such as voice telecommunication, via a packet switched network and in particular to an apparatus and method for voice, facsimile and multimedia over Internet Protocol (IP) communications components.
2. Description of the Related Art
Voice telecommunications has traditionally been conducted via dedicated telephone networks utilizing telephone switching offices and either wired or wireless connections for transmitting the voice signal between the users' telephones. Such telecommunications, which use the Public Switched Telephone Network (PSTN), may be referred to as circuit committed communications. Voice over Internet Protocol (VOIP) provides an alternative voice telecommunication means which use discrete packets digitized voice information to transmit the voice signals. The packets are transmitted either over the public Internet or within intranets.
Typical VoIP network infrastructure includes gateways, gatekeepers, proxy servers, softswitches, session border controllers, etc. Due to optimization of network resources and to particular designs, network operators may choose to integrate functionality of the separate components with one another such that multiple infrastructure components can be collocated on one physical component.
It is desirable that the VoIP network infrastructure components be designed into a network such that network operators can provide meaningful services to their customers.
The following terms are used in this disclosure:
Gateway—An entity that can bridge or serve as a “gateway” between networks. In VoIP, it typically refers to a device that can “gateway” between the traditional Public Switched Telephone Network (PSTN) and the VoIP network.
Gatekeeper—An entity that works in conjunction with the gateway to determine how to handle VoIP calls. The gatekeeper can be either in the call path or play only a consultative role in every call. The gatekeeper usually only handles VoIP calls setup using the H.323 protocol.
Proxy Server—An intermediate entity, similar in functionality to the gatekeeper, that determines how to handle VoIP calls. A proxy server usually only handles VoIP calls setup using the session initiation protocol (SIP).
User Agent—An entity that can place or receive a VoIP call, usually based on the SIP protocol (session initiation protocol).
Border element—A border element is also called network edge element. This is typically where the policy definitions or the administrative control changes. Policy can be defined at virtually all layers in the seven layer open systems interconnection (OSI) model. For example, at layer three of the seven layer model policy can typically be described in terms of routing peers, advertised IP routes etc. Routers would typically act as the border elements where such policies change between networks. Network address translators (NATs) act as border elements to connect two or more non-routable address domains. Firewalls implement policy control (for layer three and above) as border elements where the administrative control changes. The application layer typically uses flows at lower layers as well (for example, in the network layer and the transport layer). Control of the application layer potentially allows control of microflows at lower layers. For example, individual media streams for SIP calls having identical layer three characteristics may be subject to different policies. Session layer border control (SBC) allows other border elements (like routers, NAT/Firewalls, and quality of service brokers) to understand these microflows and provide the appropriate policy on a more granular basis. As a stand-alone element, an SBC simply allows policy control at the application layer.
Subnet—A subnet is an IP (Internet protocol) subnetwork inside a realm
Call Peer—JA call peer is a logical grouping for calls) Call peers may be static (created by the administrator) or dynamic (created at runtime by the multi-protocol session controller). A call peer must belong to a single device and may belong to one or more call peer groups. There are two kinds of call peers: an ingress call peer and an egress call peer, as defined in the following.
Ingress Call Peer—An ingress call peer is a call peer which is associated with the incoming of a call.
Egress Call Peer—An egress call peer is a call peer which is associated outgoing of a call.
Call Peer Group—A call peer group is a (logical) grouping of call peers based on policy (business policy, for example, service level assurances or allocation of enterprise resources), for example, sites or peers.
Device—A device is a collection of call peers. A device may be static (have a fixed binding between call peers and a layer three address) or dynamic (when protocol registrations are to create the binding between call peers and layer three addresses). A dynamic device may have static or dynamic call peers. A static device only has static call peers.
Template—A template is a rule set used for dynamically managing devices and call peers, such as subnets.
IWF—SIP/H.323 Inter-working Function
A-O-R SIP—Address of Record (RFC 3261)
AAA—Authentication, Authorization and Accounting. These refer to the three functions performed for every call to authenticate a user's phone call, authorize the user to utilize resources in the network and account for the resource usage.